The present invention relates to speech transmission in a packet network and especially to transmission between a transcoder and a base station of a mobile communication network.
In a digital telephone system a speech signal is encoded in some manner before it is channel coded and sent to the radio path. In speech coding digitalized speech is processed frame by frame at intervals of about 20 ms by using different methods so that it results in a parameter group representing speech for each frame. This information, that is, the parameter group is channel coded and sent to the transmission path. The used speech coding algorithms are RPE-LTP (Regular Pulse Excitation LPC with Long Term Prediction) and various code excited algorithms CELP (Code Excited Linear Prediction) of which VSELP (Vector-Sum Excited Linear Prediction) should be mentioned. Encoders using varying algorithms have it in common that an encoder produces speech frames of 20 ms in duration.
In addition to actual coding, the following functions are also built in for speech processing: a) on the transmitter side Voice Activity Detection VAD with which the transmitter can be instructed to be switched on only when there is speech to be sent (Discontinuous Transmission, DTX), b) on the transmitter side the evaluation of background noise and the generation of respective noise parameters and on the reception side the generation of comfort noise in a decoder from the noise parameters, and c) acoustic echo suppression. Noise during a break makes the connection sound more pleasant than absolute silence.
In known GSM mobile telephone systems the input of a speech encoder is either a PCM signal of 13 bits from the network or an AND converted PCM of 13 bits from the audio part of the mobile station. The speech frame obtained from the output of the encoder is20 ms in duration and comprises 260 audio bits which are formed by encoding 160 of PCM-encoded speech samples. Voice activity detection (SAD) defines from the parameters in the speech frame whether the frame contains speech or not. If speech is detected, the frames transmitted to the radio path as so-called traffic frames are speech frames. After a speech burst, and at specified intervals also during the speech breaks indicated by the VAD, the traffic frames are SID frames (Silence Descriptor) containing noise parameters, in which case the receiver is able to generate from these parameters noise similar to the original noise also during breaks.
A traffic frame thus contains a speech block of 260 bits representing 20 ms of encoded speech/data or noise. Furthermore, the frame has 56 bits available for frame synchronization, speech and data indication, timing and other information, the total length of the traffic frame being 316 bits. Traffic frames in the uplink and downlink direction differ from one another only in these 56 bits.
Referring to FIG. 1, which shows a simplified view of the present GSM network from the point of view of transmission. Network Subsystem comprises a mobile service switching centre, the mobile communication network being connected via the system interface of the mobile services switching centre to other networks than Public Switched Telephone Network PSTN. Via A interface the network subsystem is connected to the base station subsystem BSS comprising base station controllers, each of which controls base stations BTS connected thereto. The interface between the base station controller and the base stations connected thereto is an Abis interface. The base stations are in radio communication with mobile stations via the radio interface. Traffic frame forming unit TRAU explained above is in FIG. 1 placed in association with the base station but is may also be situated in association with the mobile services switching centre.
The mobile services switching centre MSC is shown in a simplified way in FIG. 2. Control of the base station system BSS is one function of the mobile services switching centre in addition to a call control. The function of the switching matrix is to select, switch and separate speech/data and signaling paths passing through it in the desired way. The switching matrix switches in this way its part of the connection between a mobile subscriber and a subscriber of another network and of the connection between two mobile subscribers. The function of the Network Interworking Functions IWF 1 is to adapt the GSM network into other networks. The PCM trunk line is connected to a PBX system by a terminal circuit trunk interface 3 so that the physical interface of layer 1 between the exchange and the base station controller BSC is a line of 2 Mbit/s, that is, 32 time slots of 64 kbit/s (=2048 kbit/s). The signaling terminal 4 carries out signaling according recommendation CCITT No:7.
The functions of the base station controller indicated with reference 14 in FIG. 1 include selection of a channel between it and the mobile station, link control and channel release. It carries out mapping from the radio channel to the channel of the PCM time slot of the interconnecting line between the base station and the base station controller. The base station controller shown in a simplified way in FIG. 3 comprises terminal circuits, trunk interfaces 31 and 32 by means of which the base station controller is connected on the one hand to the mobile services switching centre over the A interface and on the other hand to the base stations over the Abis interface. Transcoder and Rate Adaptation Unit TRAU is part of the base station system BSS and it may be situated in association with the base station controller BSC as shown in FIG. 1 or also in association with the mobile services switching centre. The transcoders convert speech from one digital format to another, they for example convert the 64 kbit/s A-law PCM from the exchange over the A interface into encoded speech of 13 kbit/s to be sent to the base station line and vice versa. The rate adaptation for the data is carried out in between the rate 64 kbit/s and the rates 3.6, 6 or 12 kbit/s.
The base station controller BSC configures, allocates and supervises the circuits of 64 kbit/s in the direction of the base station. It also controls the switching circuits of the base station by means of the PCM signaling link and allows the circuits of 64 kbit/s to be used efficiently, that is, a switch at the base station, which the base station controller controls, switches transmitter/receivers to PCM links. This switch hence operates as a drop/insert multiplexer which drops a PCM time slot for the transmitter of data or inserts a reception time slot to a PCM time slot of data or links the PCM time slots forwards to other base stations. The base station controller thus sets up and releases connections to the mobile station. The connections from the base stations to the PCM line/lines over the A interface and the procedure in the opposite way are multiplexed in a switching matrix 33.
The physical interface of layer 1 between the exchange and the base station controller BSC is a line of 2 Mbit/s, that is, 32 time slots of 64 kbit/s (=2048 kbit/s). The base station is totally controlled by the base station controller BSC and it mainly contains transmitter/receivers TRX which implement the radio interface towards the mobile station. Four full rate traffic channels via the radio interface can be multiplexed into one PCM channel of 64 kbit/s between the base station controller and the base station, in which case the rate of the speech/data channel is in this interval 16 kbit/s. In that case one PCM link of 64 kbit/s can transmit four speech/data connections.
FIG. 1 illustrates the transmission rates used in the GSM for a channel. The mobile station sends speech or data information over the radio interface on the radio channel as traffic frames. A base station 13 receives the information and transmits it to the time slot of 64 kbit/s of the PCM line. The other three traffic channels of the same carrier wave are also inserted in the same time slot, that is, the channel, so that the transmission rate for a connection is 16 kbit/s. In a base station controller 14 the transcoder/rate adaptation unit TRAU converts the rate 16 kbit/s of the encoded digital information into the rate 64 kbit/s and at this rate the data is transmitted to the mobile services switching centre after which, subsequent to possibly necessary modulation and rate modification, the information is transmitted to some other network.
In accordance with the foregoing explanation, the base station controller selects the circuits with which the connection is provided between it and the transmitter/receivers of the base station. The radio channel (TDMA time slot) and the PCM time slot of the line between the base station and the base station controller has during the connection a one-to-one correspondence, that is, in the uplink direction the information of a specified time slot of a specified carrier wave is always inserted in the same PCM channel of 16 kbit/s and correspondingly, in the downlink direction this information of the PCM channel is transmitted to the same TDMA time slot.
The base station controller signals to the base station which base station of the TDMA time slot has to be bound to which PCM channel. In that way the base station controller alone allocates the channel through the Abis interface and radio interface as far as the mobile station. When the base station has allocated a channel as far as the mobile station, a mobile services switching centre 15 selects the circuit with which the connection between the mobile services switching centre and the base station controller/TRAU is generated, that is, the circuits towards the A interface of the exchange and the base station controller. At the end the generated links are connected to each other.
Data transmission standard ATM (Asynchronous Transfer Mode) has been introduced for combinations of narrow band and broad band implementations and for transmission of packets and signaling. It is a connection-oriented packet switching technique which the international telecommunication standardization organization ITU-T has chosen as an implementation technique of Broadband Integrated Services Digital Network (B-ISDN). In the ATM data is packed in frames which comprise several packets of a constant length known as cells. A cell is 53 bytes in length and a cell comprises a header of 5 bytes in length and 48 bytes have been reserved for a payload. When an ATM segment is sent, each of its cells can be directed to different directions as the destination of the cell is indicated in the header.
ATM technique is best suited for use in broadband networks, especially in transmission networks using fibre optics. It is therefore probable that in the mobile communication network the present PCM technique using trunk lines of 2 Mbit/s, which the mobile operator has often hired from other teleoperators, will be replaced with ATM technique. It is necessary to operate in this way especially if the transmission capacity of the radio path is increased so much that the present PCM connection alone is no longer sufficient. In that case the data transmission capacity and the rate of the mobile communication network would increase considerably. It is also possible that the premises where the new base station is positioned already have an existing ATM connection in which case it would be tempting to use it.
Speech transmission in ATM cells has become a problem. In present circuit-switched connections speech transmission is very fast and delays hardly ever cause problems. Instead, it has become a problem how to manage transmission delays when various audio signals to the network from any of the several input points are transmitted by the ATM technique to any of the numerous output points of the network. It is a particular problem how to transmit audio signals converted as PCM encoded and multiplexed in PCM devices between the nodes of the network and across the network, which network contains ATM transfer devices and exchanges.
The solutions given to this problem are at least the following a) use of microcells, b) incomplete filling of cells, and c) emulation of circuit switching.
When micro cells are used, several speech channels are multiplexed for transporting one ATM cell. It is a problem with the micro cell technique is that the ATM cell is no longer the basic unit of switching in which case ordinary ATM switching devices cannot be used to switch speech channels but special arrangements and devices are needed for releasing speech channels inside the microcells. In incomplete filling of ATM cells, the payload of the cell is left incomplete. In this way the capacity is underused but it has to be done if delays are to be avoided. In emulation of circuit switching, information moving on the PCM line of 2 Mbit/s is transmitted transparently in one ATM cell flux. A disadvantage of this method is that transmission capacity is always reserved regardless of whether there are calls to be transmitted or not, wherefore transmission of empty cells cannot be avoided. Another disadvantage is that speech channels of the connection of the point-to-point nature cannot be connected/switched with ATM equipments inside the network into different directions.
Patent Application WO 94/11975 discloses a method, a telecommunication network and a switching system for transmitting several PCM encoded speech channels through the ATM network. The method includes features of steps a and c mentioned above. According to the application several speech channels assigned for the same output node of the ATM network are packed in one ATM cell, whereby sound and narrowband data channels are transmitted in these cells which are transmitted at a reproducing rate which is the same or an integral part of the reproducing rate of sound-containing PCM signal. Cells are transmitted in the network between the input node and the output node via virtual circuits maintaining a constant rate. When there are no great changes in the traffic so that permanent virtual paths need to be added between two nodes or deleted, the switching system carries out a simple function: a frame of PCM samples at the input point of 125 microseconds in duration, inserted in one ATM cell is routed through the network to the output node which means that cells are sent at intervals of 125 milliseconds. One PCM sample comprises one byte, wherefore 48 speech channels at the maximum can be transmitted in one cell. If the capacity of the PCM channel is more than 64 kbit/s, e.g. 384 kbit/s, more bytes are used of the cell for one channel, for example 6 bytes.
None of the above explained methods is as such suitable when the transmission of audio information of PCM channel between the base station and TRAU is replaced with the ATM connection in order that speech information can be transmitted, when required, directly from one base station to another without the connection passing through the TRAU or the mobile services switching centre as in the present GSM system.
The object of the present invention is thus a method by means of which speech comprising speech frames generated from a PCM encoded speech signal of the speech encoder can be transmitted without a disadvantageous delay in a packet network, such as the ATM network. A particular object is a method by means of which speech can be transmitted in packet mode between a base station and a TRAU and two base stations in the mobile communication system.
The object is attained with the method and the digital mobile communication network according to claims 1 and 4.
A digitalized speech signal is converted frame by frame in a speech encoder into a parameter group which is inserted in a traffic frame. A traffic frame may be a speech frame as such but mostly additional bits are needed for different purposes for the transmission in which case the length of the frame is greater than the length of a mere speech frame. The provided traffic frame as a whole is inserted immediately in the payload part of the data packet and the data packet is sent via the transmission network to the destination. At the destination the traffic frame is separated from the payload of the received packet and it is passed to the speech decoder for restoring the original digitalized speech signal. According to the preferred embodiment, the transmission network is an ATM network in which case the packet is an ATM cell.
According to claim 4 the transmission link is provided by a packet network which transmits a call between base stations and a transcoder of the mobile communication network.
Other information may be inserted, when desired, in the payload portion of the packet/cell that has remained free, for example information needed for identifying the call in the uplink direction which can be used for directing ATM cells associated with the call in the MSC.
The advantages of the invention are first of all reduced transmission delay in the network and secondly, the transmission of one call in one ATM cell enables ATM switching of the cells and thus directing the call into the desired destination. This results in a genuine ATM telephone network. Thirdly, the transmission of the call in one ATM cell makes it possible that after the call has been terminated, the transmission of the cell also ends, which is contrary to when circuit switching is emulated. Fourthly, the cells need to be sent during breaks in speech only when noise parameters are transmitted. Transmission capacity is thus released for other use during breaks.